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<A href="http://en.wikipedia.org/wiki/Digital_Signal_1">http://en.wikipedia.org/wiki/Digital_Signal_1</A><BR>
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I believe it's BRI lines that uses 56kbps and your right that SS7 on BRI have some usage in US.<BR>
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Jan <BR>
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> Date: Wed, 7 Dec 2011 20:12:22 -0200<BR>> From: marcelo@m2j.com.br<BR>> To: asterisk-ss7@lists.digium.com<BR>> Subject: Re: [asterisk-ss7] SS7 + T1 on Asterisk?<BR>> <BR>> Typically T1 (american) signaling ss7 links run at 56kbps instead of 64kbps.<BR>> If your switch can run 64kbps links over a T1 timeslot, than the only <BR>> remaining variable is ITU versus ANSI ISUP. They are incompatible <BR>> (different message formats due to different network address sizes and <BR>> other details).<BR>> We use ITU ISUP all over the place without trouble. If the switch can do <BR>> 64kbps links and ITU ISUP, then you should be able to use all existing <BR>> E1 direct connection samples (without STP), except for the obvious E1=31 <BR>> timeslots while T1=24 timeslots difference..<BR>> ANSI might work. I won't go there because I have zero experience with <BR>> ANSI SS7/ISUP (stability wise).<BR>> With 2 T1 and a single signaling link it should allow for 47 voice <BR>> channels and one signaling link.<BR>> <BR>> Search for libss7 ansi 56kbps for the most difficult scenario. But if <BR>> you can do ITU ISUP + 64kbps links, I would suggest that instead.<BR>> We hardly see people talking about ANSI ISUP setups on this list, so it <BR>> could be far less stable (at least it seems to get less usage).<BR>> <BR>> On 12/07/11 16:25, Matt wrote:<BR>> > In this case, our supplier is ourselves. We have a DMS100, but the<BR>> > switch guy is someone other than myself - I am the IP guy.<BR>> ><BR>> > So basically if I understand you properly, I should be able to do the<BR>> > SS7+T1 and get proper operation, provided the configuration on both<BR>> > sides is right.<BR>> ><BR>> > On Wed, Dec 7, 2011 at 1:06 PM, Marcelo Pacheco<marcelo@m2j.com.br> wrote:<BR>> >> If the DMS100 switch can talk signalling directly with Asterisk, without an<BR>> >> STP, it should be possible to use a single timeslot for ss7 signalling, so<BR>> >> with 2 T1 you could have 47 voice calls and one signalling channel. This is<BR>> >> common with E1 setups. Also with E1 its common for a timeslot to be<BR>> >> statically switched over to an STP (semi permanent call), allowing for<BR>> >> access to the signaling network without a dedicated physically separate<BR>> >> signaling link, but that's not usual in T1 land.<BR>> >><BR>> >> But what you ask is technically possible... However its important to<BR>> >> PROPERLY LEARN SS7 terms to be able to communicate with your supplier.<BR>> >> SS7 is a CARRIER LEVEL PROTOCOL. However people insist on winging it without<BR>> >> proper training.<BR>> >> Its like trying to become a backbone internet provider without properly<BR>> >> learning inter and intra network routing protocols (like BGP and OSPF).<BR>> >><BR>> >> If you knew the general SS7/ISUP knowledge, you could quickly find the<BR>> >> information you're looking for on Google.<BR>> >><BR>> >> PS: I live in E1 land... I'm just quoting information from the top of my<BR>> >> head. I have no need for T1+SS7. E1+SS7 is a little simpler with Asterisk<BR>> >> than T1+SS7 due to 56kbps data links, ANSI ISUP/SS7 and some other quirks.<BR>> >><BR>> >> Good luck. You'll need it.<BR>> >><BR>> >><BR>> >> On 12/07/11 14:47, Matt wrote:<BR>> >>> If I were to get a 2 span T1 card for Asterisk... and connect it to a<BR>> >>> Nortel DMS100... can I run call traffic over the T1 and run SS7<BR>> >>> signaling FOR the T1 over the other port?<BR>> >>><BR>> >>> Is there documentation on doing this anywhere?<BR>> >>><BR>> >>> --<BR>> >>> _____________________________________________________________________<BR>> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>> >>><BR>> >>> asterisk-ss7 mailing list<BR>> >>> To UNSUBSCRIBE or update options visit:<BR>> >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR>> >>><BR>> >><BR>> >> --<BR>> >> _____________________________________________________________________<BR>> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>> >><BR>> >> asterisk-ss7 mailing list<BR>> >> To UNSUBSCRIBE or update options visit:<BR>> >> http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR>> > --<BR>> > _____________________________________________________________________<BR>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>> ><BR>> > asterisk-ss7 mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR>> ><BR>> <BR>> <BR>> --<BR>> _____________________________________________________________________<BR>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>> <BR>> asterisk-ss7 mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR></DIV> </div></body>
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